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wu@webrtc.org364f2042013-11-20 21:49:41 +00001/*
kjellanderb24317b2016-02-10 07:54:43 -08002 * Copyright 2013 The WebRTC project authors. All Rights Reserved.
wu@webrtc.org364f2042013-11-20 21:49:41 +00003 *
kjellanderb24317b2016-02-10 07:54:43 -08004 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
wu@webrtc.org364f2042013-11-20 21:49:41 +00009 */
10
kwiberg0eb15ed2015-12-17 03:04:15 -080011#include <utility>
12
Henrik Kjellander15583c12016-02-10 10:53:12 +010013#include "webrtc/api/test/fakeperiodicvideocapturer.h"
Henrik Boströmd79599d2016-06-01 13:58:50 +020014#include "webrtc/api/test/fakertccertificategenerator.h"
Henrik Kjellander15583c12016-02-10 10:53:12 +010015#include "webrtc/api/test/mockpeerconnectionobservers.h"
16#include "webrtc/api/test/peerconnectiontestwrapper.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000017#include "webrtc/base/gunit.h"
Taylor Brandstettera1c30352016-05-13 08:15:11 -070018#include "webrtc/p2p/base/fakeportallocator.h"
wu@webrtc.org364f2042013-11-20 21:49:41 +000019
20static const char kStreamLabelBase[] = "stream_label";
21static const char kVideoTrackLabelBase[] = "video_track";
22static const char kAudioTrackLabelBase[] = "audio_track";
buildbot@webrtc.org3e01e0b2014-05-13 17:54:10 +000023static const int kMaxWait = 10000;
wu@webrtc.org364f2042013-11-20 21:49:41 +000024static const int kTestAudioFrameCount = 3;
25static const int kTestVideoFrameCount = 3;
26
27using webrtc::FakeConstraints;
28using webrtc::FakeVideoTrackRenderer;
29using webrtc::IceCandidateInterface;
30using webrtc::MediaConstraintsInterface;
31using webrtc::MediaStreamInterface;
32using webrtc::MockSetSessionDescriptionObserver;
33using webrtc::PeerConnectionInterface;
34using webrtc::SessionDescriptionInterface;
35using webrtc::VideoTrackInterface;
36
37void PeerConnectionTestWrapper::Connect(PeerConnectionTestWrapper* caller,
38 PeerConnectionTestWrapper* callee) {
39 caller->SignalOnIceCandidateReady.connect(
40 callee, &PeerConnectionTestWrapper::AddIceCandidate);
41 callee->SignalOnIceCandidateReady.connect(
42 caller, &PeerConnectionTestWrapper::AddIceCandidate);
43
44 caller->SignalOnSdpReady.connect(
45 callee, &PeerConnectionTestWrapper::ReceiveOfferSdp);
46 callee->SignalOnSdpReady.connect(
47 caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp);
48}
49
danilchape9021a32016-05-17 01:52:02 -070050PeerConnectionTestWrapper::PeerConnectionTestWrapper(
51 const std::string& name,
52 rtc::Thread* network_thread,
53 rtc::Thread* worker_thread)
54 : name_(name),
55 network_thread_(network_thread),
56 worker_thread_(worker_thread) {}
wu@webrtc.org364f2042013-11-20 21:49:41 +000057
58PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {}
59
60bool PeerConnectionTestWrapper::CreatePc(
zhihuang9763d562016-08-05 11:14:50 -070061 const MediaConstraintsInterface* constraints,
62 const webrtc::PeerConnectionInterface::RTCConfiguration& config) {
kwibergd1fe2812016-04-27 06:47:29 -070063 std::unique_ptr<cricket::PortAllocator> port_allocator(
danilchape9021a32016-05-17 01:52:02 -070064 new cricket::FakePortAllocator(network_thread_, nullptr));
wu@webrtc.org364f2042013-11-20 21:49:41 +000065
deadbeefee8c6d32015-08-13 14:27:18 -070066 fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
wu@webrtc.org364f2042013-11-20 21:49:41 +000067 if (fake_audio_capture_module_ == NULL) {
68 return false;
69 }
70
71 peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
danilchape9021a32016-05-17 01:52:02 -070072 network_thread_, worker_thread_, rtc::Thread::Current(),
73 fake_audio_capture_module_, NULL, NULL);
wu@webrtc.org364f2042013-11-20 21:49:41 +000074 if (!peer_connection_factory_) {
75 return false;
76 }
77
Henrik Boströmd79599d2016-06-01 13:58:50 +020078 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator(
79 rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeRTCCertificateGenerator()
kwibergd1fe2812016-04-27 06:47:29 -070080 : nullptr);
Henrik Boströmd79599d2016-06-01 13:58:50 +020081 peer_connection_ = peer_connection_factory_->CreatePeerConnection(
82 config, constraints, std::move(port_allocator), std::move(cert_generator),
83 this);
wu@webrtc.org364f2042013-11-20 21:49:41 +000084
85 return peer_connection_.get() != NULL;
86}
87
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000088rtc::scoped_refptr<webrtc::DataChannelInterface>
jiayl@webrtc.org1a6c6282014-06-12 21:59:29 +000089PeerConnectionTestWrapper::CreateDataChannel(
90 const std::string& label,
91 const webrtc::DataChannelInit& init) {
92 return peer_connection_->CreateDataChannel(label, &init);
93}
94
Taylor Brandstetter98cde262016-05-31 13:02:21 -070095void PeerConnectionTestWrapper::OnAddStream(
96 rtc::scoped_refptr<MediaStreamInterface> stream) {
wu@webrtc.org364f2042013-11-20 21:49:41 +000097 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
98 << ": OnAddStream";
99 // TODO(ronghuawu): support multiple streams.
100 if (stream->GetVideoTracks().size() > 0) {
101 renderer_.reset(new FakeVideoTrackRenderer(stream->GetVideoTracks()[0]));
102 }
103}
104
105void PeerConnectionTestWrapper::OnIceCandidate(
106 const IceCandidateInterface* candidate) {
107 std::string sdp;
108 EXPECT_TRUE(candidate->ToString(&sdp));
109 // Give the user a chance to modify sdp for testing.
110 SignalOnIceCandidateCreated(&sdp);
111 SignalOnIceCandidateReady(candidate->sdp_mid(), candidate->sdp_mline_index(),
112 sdp);
113}
114
jiayl@webrtc.org1a6c6282014-06-12 21:59:29 +0000115void PeerConnectionTestWrapper::OnDataChannel(
Taylor Brandstetter98cde262016-05-31 13:02:21 -0700116 rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) {
jiayl@webrtc.org1a6c6282014-06-12 21:59:29 +0000117 SignalOnDataChannel(data_channel);
118}
119
wu@webrtc.org364f2042013-11-20 21:49:41 +0000120void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +0000121 // This callback should take the ownership of |desc|.
kwibergd1fe2812016-04-27 06:47:29 -0700122 std::unique_ptr<SessionDescriptionInterface> owned_desc(desc);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000123 std::string sdp;
124 EXPECT_TRUE(desc->ToString(&sdp));
125
126 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
127 << ": " << desc->type() << " sdp created: " << sdp;
128
129 // Give the user a chance to modify sdp for testing.
130 SignalOnSdpCreated(&sdp);
131
132 SetLocalDescription(desc->type(), sdp);
133
134 SignalOnSdpReady(sdp);
135}
136
137void PeerConnectionTestWrapper::CreateOffer(
138 const MediaConstraintsInterface* constraints) {
139 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
140 << ": CreateOffer.";
141 peer_connection_->CreateOffer(this, constraints);
142}
143
144void PeerConnectionTestWrapper::CreateAnswer(
145 const MediaConstraintsInterface* constraints) {
146 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
147 << ": CreateAnswer.";
148 peer_connection_->CreateAnswer(this, constraints);
149}
150
151void PeerConnectionTestWrapper::ReceiveOfferSdp(const std::string& sdp) {
152 SetRemoteDescription(SessionDescriptionInterface::kOffer, sdp);
153 CreateAnswer(NULL);
154}
155
156void PeerConnectionTestWrapper::ReceiveAnswerSdp(const std::string& sdp) {
157 SetRemoteDescription(SessionDescriptionInterface::kAnswer, sdp);
158}
159
160void PeerConnectionTestWrapper::SetLocalDescription(const std::string& type,
161 const std::string& sdp) {
162 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
163 << ": SetLocalDescription " << type << " " << sdp;
164
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000165 rtc::scoped_refptr<MockSetSessionDescriptionObserver>
166 observer(new rtc::RefCountedObject<
wu@webrtc.org364f2042013-11-20 21:49:41 +0000167 MockSetSessionDescriptionObserver>());
168 peer_connection_->SetLocalDescription(
169 observer, webrtc::CreateSessionDescription(type, sdp, NULL));
170}
171
172void PeerConnectionTestWrapper::SetRemoteDescription(const std::string& type,
173 const std::string& sdp) {
174 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
175 << ": SetRemoteDescription " << type << " " << sdp;
176
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000177 rtc::scoped_refptr<MockSetSessionDescriptionObserver>
178 observer(new rtc::RefCountedObject<
wu@webrtc.org364f2042013-11-20 21:49:41 +0000179 MockSetSessionDescriptionObserver>());
180 peer_connection_->SetRemoteDescription(
181 observer, webrtc::CreateSessionDescription(type, sdp, NULL));
182}
183
184void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid,
185 int sdp_mline_index,
186 const std::string& candidate) {
kwibergd1fe2812016-04-27 06:47:29 -0700187 std::unique_ptr<webrtc::IceCandidateInterface> owned_candidate(
sergeyu@chromium.org5bc25c42013-12-05 00:24:06 +0000188 webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL));
189 EXPECT_TRUE(peer_connection_->AddIceCandidate(owned_candidate.get()));
wu@webrtc.org364f2042013-11-20 21:49:41 +0000190}
191
192void PeerConnectionTestWrapper::WaitForCallEstablished() {
193 WaitForConnection();
194 WaitForAudio();
195 WaitForVideo();
196}
197
198void PeerConnectionTestWrapper::WaitForConnection() {
199 EXPECT_TRUE_WAIT(CheckForConnection(), kMaxWait);
200 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
201 << ": Connected.";
202}
203
204bool PeerConnectionTestWrapper::CheckForConnection() {
205 return (peer_connection_->ice_connection_state() ==
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000206 PeerConnectionInterface::kIceConnectionConnected) ||
207 (peer_connection_->ice_connection_state() ==
208 PeerConnectionInterface::kIceConnectionCompleted);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000209}
210
211void PeerConnectionTestWrapper::WaitForAudio() {
212 EXPECT_TRUE_WAIT(CheckForAudio(), kMaxWait);
213 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
214 << ": Got enough audio frames.";
215}
216
217bool PeerConnectionTestWrapper::CheckForAudio() {
218 return (fake_audio_capture_module_->frames_received() >=
219 kTestAudioFrameCount);
220}
221
222void PeerConnectionTestWrapper::WaitForVideo() {
223 EXPECT_TRUE_WAIT(CheckForVideo(), kMaxWait);
224 LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
225 << ": Got enough video frames.";
226}
227
228bool PeerConnectionTestWrapper::CheckForVideo() {
229 if (!renderer_) {
230 return false;
231 }
232 return (renderer_->num_rendered_frames() >= kTestVideoFrameCount);
233}
234
235void PeerConnectionTestWrapper::GetAndAddUserMedia(
236 bool audio, const webrtc::FakeConstraints& audio_constraints,
237 bool video, const webrtc::FakeConstraints& video_constraints) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000238 rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
wu@webrtc.org364f2042013-11-20 21:49:41 +0000239 GetUserMedia(audio, audio_constraints, video, video_constraints);
perkj@webrtc.orgc2dd5ee2014-11-04 11:31:29 +0000240 EXPECT_TRUE(peer_connection_->AddStream(stream));
wu@webrtc.org364f2042013-11-20 21:49:41 +0000241}
242
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000243rtc::scoped_refptr<webrtc::MediaStreamInterface>
wu@webrtc.org364f2042013-11-20 21:49:41 +0000244 PeerConnectionTestWrapper::GetUserMedia(
245 bool audio, const webrtc::FakeConstraints& audio_constraints,
246 bool video, const webrtc::FakeConstraints& video_constraints) {
247 std::string label = kStreamLabelBase +
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000248 rtc::ToString<int>(
wu@webrtc.org364f2042013-11-20 21:49:41 +0000249 static_cast<int>(peer_connection_->local_streams()->count()));
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000250 rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
wu@webrtc.org364f2042013-11-20 21:49:41 +0000251 peer_connection_factory_->CreateLocalMediaStream(label);
252
253 if (audio) {
254 FakeConstraints constraints = audio_constraints;
255 // Disable highpass filter so that we can get all the test audio frames.
256 constraints.AddMandatory(
257 MediaConstraintsInterface::kHighpassFilter, false);
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000258 rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
wu@webrtc.org364f2042013-11-20 21:49:41 +0000259 peer_connection_factory_->CreateAudioSource(&constraints);
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000260 rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
wu@webrtc.org364f2042013-11-20 21:49:41 +0000261 peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
262 source));
263 stream->AddTrack(audio_track);
264 }
265
266 if (video) {
267 // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
268 FakeConstraints constraints = video_constraints;
269 constraints.SetMandatoryMaxFrameRate(10);
270
perkja3ede6c2016-03-08 01:27:48 +0100271 rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
wu@webrtc.org364f2042013-11-20 21:49:41 +0000272 peer_connection_factory_->CreateVideoSource(
273 new webrtc::FakePeriodicVideoCapturer(), &constraints);
274 std::string videotrack_label = label + kVideoTrackLabelBase;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000275 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
wu@webrtc.org364f2042013-11-20 21:49:41 +0000276 peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
277
278 stream->AddTrack(video_track);
279 }
280 return stream;
281}